d056fa046c
sound cards with optional pseudo-multichannel playback.
It's based on snd_emu10k1 sound driver. Single channel version is available
from audio/emu10kx port since some time.
The two new ALSA header files (GPLed), which contain Audigy 2 ("p16v") and
Audigy 2 Value ("p17v") specific interfaces, are latest versions from ALSA
Mercurial repository.
This is not connected to the build yet.
Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
302 lines
15 KiB
C
302 lines
15 KiB
C
/*-
|
|
* Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk>
|
|
* Driver p16v chips
|
|
* Version: 0.21
|
|
*
|
|
* FEATURES currently supported:
|
|
* Output fixed at S32_LE, 2 channel to hw:0,0
|
|
* Rates: 44.1, 48, 96, 192.
|
|
*
|
|
* Changelog:
|
|
* 0.8
|
|
* Use separate card based buffer for periods table.
|
|
* 0.9
|
|
* Use 2 channel output streams instead of 8 channel.
|
|
* (8 channel output streams might be good for ASIO type output)
|
|
* Corrected speaker output, so Front -> Front etc.
|
|
* 0.10
|
|
* Fixed missed interrupts.
|
|
* 0.11
|
|
* Add Sound card model number and names.
|
|
* Add Analog volume controls.
|
|
* 0.12
|
|
* Corrected playback interrupts. Now interrupt per period, instead of half period.
|
|
* 0.13
|
|
* Use single trigger for multichannel.
|
|
* 0.14
|
|
* Mic capture now works at fixed: S32_LE, 96000Hz, Stereo.
|
|
* 0.15
|
|
* Force buffer_size / period_size == INTEGER.
|
|
* 0.16
|
|
* Update p16v.c to work with changed alsa api.
|
|
* 0.17
|
|
* Update p16v.c to work with changed alsa api. Removed boot_devs.
|
|
* 0.18
|
|
* Merging with snd-emu10k1 driver.
|
|
* 0.19
|
|
* One stereo channel at 24bit now works.
|
|
* 0.20
|
|
* Added better register defines.
|
|
* 0.21
|
|
* Split from p16v.c
|
|
*
|
|
*
|
|
* BUGS:
|
|
* Some stability problems when unloading the snd-p16v kernel module.
|
|
* --
|
|
*
|
|
* TODO:
|
|
* SPDIF out.
|
|
* Find out how to change capture sample rates. E.g. To record SPDIF at 48000Hz.
|
|
* Currently capture fixed at 48000Hz.
|
|
*
|
|
* --
|
|
* GENERAL INFO:
|
|
* Model: SB0240
|
|
* P16V Chip: CA0151-DBS
|
|
* Audigy 2 Chip: CA0102-IAT
|
|
* AC97 Codec: STAC 9721
|
|
* ADC: Philips 1361T (Stereo 24bit)
|
|
* DAC: CS4382-K (8-channel, 24bit, 192Khz)
|
|
*
|
|
* This code was initally based on code from ALSA's emu10k1x.c which is:
|
|
* Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
|
|
*
|
|
* This program is free software; you can redistribute it and/or modify
|
|
* it under the terms of the GNU General Public License as published by
|
|
* the Free Software Foundation; either version 2 of the License, or
|
|
* (at your option) any later version.
|
|
*
|
|
* This program is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
|
|
* GNU General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU General Public License
|
|
* along with this program; if not, write to the Free Software
|
|
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
|
*
|
|
*/
|
|
|
|
/* $FreeBSD$ */
|
|
|
|
/********************************************************************************************************/
|
|
/* Audigy2 P16V pointer-offset register set, accessed through the PTR2 and DATA2 registers */
|
|
/********************************************************************************************************/
|
|
|
|
/* The sample rate of the SPDIF outputs is set by modifying a register in the EMU10K2 PTR register A_SPDIF_SAMPLERATE.
|
|
* The sample rate is also controlled by the same registers that control the rate of the EMU10K2 sample rate converters.
|
|
*/
|
|
|
|
/* Initally all registers from 0x00 to 0x3f have zero contents. */
|
|
#define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */
|
|
/* One list entry: 4 bytes for DMA address,
|
|
* 4 bytes for period_size << 16.
|
|
* One list entry is 8 bytes long.
|
|
* One list entry for each period in the buffer.
|
|
*/
|
|
#define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */
|
|
#define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */
|
|
#define PLAYBACK_UNKNOWN3 0x03 /* Not used */
|
|
#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA addresss */
|
|
#define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size. win2000 uses 0x04000000 */
|
|
#define PLAYBACK_POINTER 0x06 /* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */
|
|
#define PLAYBACK_FIFO_END_ADDRESS 0x07 /* Playback FIFO end address */
|
|
#define PLAYBACK_FIFO_POINTER 0x08 /* Playback FIFO pointer and number of valid sound samples in cache */
|
|
#define PLAYBACK_UNKNOWN9 0x09 /* Not used */
|
|
#define CAPTURE_DMA_ADDR 0x10 /* Capture DMA address */
|
|
#define CAPTURE_BUFFER_SIZE 0x11 /* Capture buffer size */
|
|
#define CAPTURE_POINTER 0x12 /* Capture buffer pointer. Sample currently in ADC */
|
|
#define CAPTURE_FIFO_POINTER 0x13 /* Capture FIFO pointer and number of valid sound samples in cache */
|
|
#define CAPTURE_P16V_VOLUME1 0x14 /* Low: Capture volume 0xXXXX3030 */
|
|
#define CAPTURE_P16V_VOLUME2 0x15 /* High:Has no effect on capture volume */
|
|
#define CAPTURE_P16V_SOURCE 0x16 /* P16V source select. Set to 0x0700E4E5 for AC97 CAPTURE */
|
|
/* [0:1] Capture input 0 channel select. 0 = Capture output 0.
|
|
* 1 = Capture output 1.
|
|
* 2 = Capture output 2.
|
|
* 3 = Capture output 3.
|
|
* [3:2] Capture input 1 channel select. 0 = Capture output 0.
|
|
* 1 = Capture output 1.
|
|
* 2 = Capture output 2.
|
|
* 3 = Capture output 3.
|
|
* [5:4] Capture input 2 channel select. 0 = Capture output 0.
|
|
* 1 = Capture output 1.
|
|
* 2 = Capture output 2.
|
|
* 3 = Capture output 3.
|
|
* [7:6] Capture input 3 channel select. 0 = Capture output 0.
|
|
* 1 = Capture output 1.
|
|
* 2 = Capture output 2.
|
|
* 3 = Capture output 3.
|
|
* [9:8] Playback input 0 channel select. 0 = Play output 0.
|
|
* 1 = Play output 1.
|
|
* 2 = Play output 2.
|
|
* 3 = Play output 3.
|
|
* [11:10] Playback input 1 channel select. 0 = Play output 0.
|
|
* 1 = Play output 1.
|
|
* 2 = Play output 2.
|
|
* 3 = Play output 3.
|
|
* [13:12] Playback input 2 channel select. 0 = Play output 0.
|
|
* 1 = Play output 1.
|
|
* 2 = Play output 2.
|
|
* 3 = Play output 3.
|
|
* [15:14] Playback input 3 channel select. 0 = Play output 0.
|
|
* 1 = Play output 1.
|
|
* 2 = Play output 2.
|
|
* 3 = Play output 3.
|
|
* [19:16] Playback mixer output enable. 1 bit per channel.
|
|
* [23:20] Capture mixer output enable. 1 bit per channel.
|
|
* [26:24] FX engine channel capture 0 = 0x60-0x67.
|
|
* 1 = 0x68-0x6f.
|
|
* 2 = 0x70-0x77.
|
|
* 3 = 0x78-0x7f.
|
|
* 4 = 0x80-0x87.
|
|
* 5 = 0x88-0x8f.
|
|
* 6 = 0x90-0x97.
|
|
* 7 = 0x98-0x9f.
|
|
* [31:27] Not used.
|
|
*/
|
|
|
|
/* 0x1 = capture on.
|
|
* 0x100 = capture off.
|
|
* 0x200 = capture off.
|
|
* 0x1000 = capture off.
|
|
*/
|
|
#define CAPTURE_RATE_STATUS 0x17 /* Capture sample rate. Read only */
|
|
/* [15:0] Not used.
|
|
* [18:16] Channel 0 Detected sample rate. 0 - 44.1khz
|
|
* 1 - 48 khz
|
|
* 2 - 96 khz
|
|
* 3 - 192 khz
|
|
* 7 - undefined rate.
|
|
* [19] Channel 0. 1 - Valid, 0 - Not Valid.
|
|
* [22:20] Channel 1 Detected sample rate.
|
|
* [23] Channel 1. 1 - Valid, 0 - Not Valid.
|
|
* [26:24] Channel 2 Detected sample rate.
|
|
* [27] Channel 2. 1 - Valid, 0 - Not Valid.
|
|
* [30:28] Channel 3 Detected sample rate.
|
|
* [31] Channel 3. 1 - Valid, 0 - Not Valid.
|
|
*/
|
|
/* 0x18 - 0x1f unused */
|
|
#define PLAYBACK_LAST_SAMPLE 0x20 /* The sample currently being played. Read only */
|
|
/* 0x21 - 0x3f unused */
|
|
#define BASIC_INTERRUPT 0x40 /* Used by both playback and capture interrupt handler */
|
|
/* Playback (0x1<<channel_id) Don't touch high 16bits. */
|
|
/* Capture (0x100<<channel_id). not tested */
|
|
/* Start Playback [3:0] (one bit per channel)
|
|
* Start Capture [11:8] (one bit per channel)
|
|
* Record source select for channel 0 [18:16]
|
|
* Record source select for channel 1 [22:20]
|
|
* Record source select for channel 2 [26:24]
|
|
* Record source select for channel 3 [30:28]
|
|
* 0 - SPDIF channel.
|
|
* 1 - I2S channel.
|
|
* 2 - SRC48 channel.
|
|
* 3 - SRCMulti_SPDIF channel.
|
|
* 4 - SRCMulti_I2S channel.
|
|
* 5 - SPDIF channel.
|
|
* 6 - fxengine capture.
|
|
* 7 - AC97 capture.
|
|
*/
|
|
/* Default 41110000.
|
|
* Writing 0xffffffff hangs the PC.
|
|
* Writing 0xffff0000 -> 77770000 so it must be some sort of route.
|
|
* bit 0x1 starts DMA playback on channel_id 0
|
|
*/
|
|
/* 0x41,42 take values from 0 - 0xffffffff, but have no effect on playback */
|
|
/* 0x43,0x48 do not remember settings */
|
|
/* 0x41-45 unused */
|
|
#define WATERMARK 0x46 /* Test bit to indicate cache level usage */
|
|
/* Values it can have while playing on channel 0.
|
|
* 0000f000, 0000f004, 0000f008, 0000f00c.
|
|
* Readonly.
|
|
*/
|
|
/* 0x47-0x4f unused */
|
|
/* 0x50-0x5f Capture cache data */
|
|
#define SRCSel 0x60 /* SRCSel. Default 0x4. Bypass P16V 0x14 */
|
|
/* [0] 0 = 10K2 audio, 1 = SRC48 mixer output.
|
|
* [2] 0 = 10K2 audio, 1 = SRCMulti SPDIF mixer output.
|
|
* [4] 0 = 10K2 audio, 1 = SRCMulti I2S mixer output.
|
|
*/
|
|
/* SRC48 converts samples rates 44.1, 48, 96, 192 to 48 khz. */
|
|
/* SRCMulti converts 48khz samples rates to 44.1, 48, 96, 192 to 48. */
|
|
/* SRC48 and SRCMULTI sample rate select and output select. */
|
|
/* 0xffffffff -> 0xC0000015
|
|
* 0xXXXXXXX4 = Enable Front Left/Right
|
|
* Enable PCMs
|
|
*/
|
|
|
|
/* 0x61 -> 0x6c are Volume controls */
|
|
#define PLAYBACK_VOLUME_MIXER1 0x61 /* SRC48 Low to mixer input volume control. */
|
|
#define PLAYBACK_VOLUME_MIXER2 0x62 /* SRC48 High to mixer input volume control. */
|
|
#define PLAYBACK_VOLUME_MIXER3 0x63 /* SRCMULTI SPDIF Low to mixer input volume control. */
|
|
#define PLAYBACK_VOLUME_MIXER4 0x64 /* SRCMULTI SPDIF High to mixer input volume control. */
|
|
#define PLAYBACK_VOLUME_MIXER5 0x65 /* SRCMULTI I2S Low to mixer input volume control. */
|
|
#define PLAYBACK_VOLUME_MIXER6 0x66 /* SRCMULTI I2S High to mixer input volume control. */
|
|
#define PLAYBACK_VOLUME_MIXER7 0x67 /* P16V Low to SRCMULTI SPDIF mixer input volume control. */
|
|
#define PLAYBACK_VOLUME_MIXER8 0x68 /* P16V High to SRCMULTI SPDIF mixer input volume control. */
|
|
#define PLAYBACK_VOLUME_MIXER9 0x69 /* P16V Low to SRCMULTI I2S mixer input volume control. */
|
|
/* 0xXXXX3030 = PCM0 Volume (Front).
|
|
* 0x3030XXXX = PCM1 Volume (Center)
|
|
*/
|
|
#define PLAYBACK_VOLUME_MIXER10 0x6a /* P16V High to SRCMULTI I2S mixer input volume control. */
|
|
/* 0x3030XXXX = PCM3 Volume (Rear). */
|
|
#define PLAYBACK_VOLUME_MIXER11 0x6b /* E10K2 Low to SRC48 mixer input volume control. */
|
|
#define PLAYBACK_VOLUME_MIXER12 0x6c /* E10K2 High to SRC48 mixer input volume control. */
|
|
|
|
#define SRC48_ENABLE 0x6d /* SRC48 input audio enable */
|
|
/* SRC48 converts samples rates 44.1, 48, 96, 192 to 48 khz. */
|
|
/* [23:16] The corresponding P16V channel to SRC48 enabled if == 1.
|
|
* [31:24] The corresponding E10K2 channel to SRC48 enabled.
|
|
*/
|
|
#define SRCMULTI_ENABLE 0x6e /* SRCMulti input audio enable. Default 0xffffffff */
|
|
/* SRCMulti converts 48khz samples rates to 44.1, 48, 96, 192 to 48. */
|
|
/* [7:0] The corresponding P16V channel to SRCMulti_I2S enabled if == 1.
|
|
* [15:8] The corresponding E10K2 channel to SRCMulti I2S enabled.
|
|
* [23:16] The corresponding P16V channel to SRCMulti SPDIF enabled.
|
|
* [31:24] The corresponding E10K2 channel to SRCMulti SPDIF enabled.
|
|
*/
|
|
/* Bypass P16V 0xff00ff00
|
|
* Bitmap. 0 = Off, 1 = On.
|
|
* P16V playback outputs:
|
|
* 0xXXXXXXX1 = PCM0 Left. (Front)
|
|
* 0xXXXXXXX2 = PCM0 Right.
|
|
* 0xXXXXXXX4 = PCM1 Left. (Center/LFE)
|
|
* 0xXXXXXXX8 = PCM1 Right.
|
|
* 0xXXXXXX1X = PCM2 Left. (Unknown)
|
|
* 0xXXXXXX2X = PCM2 Right.
|
|
* 0xXXXXXX4X = PCM3 Left. (Rear)
|
|
* 0xXXXXXX8X = PCM3 Right.
|
|
*/
|
|
#define AUDIO_OUT_ENABLE 0x6f /* Default: 000100FF */
|
|
/* [3:0] Does something, but not documented. Probably capture enable.
|
|
* [7:4] Playback channels enable. not documented.
|
|
* [16] AC97 output enable if == 1
|
|
* [30] 0 = SRCMulti_I2S input from fxengine 0x68-0x6f.
|
|
* 1 = SRCMulti_I2S input from SRC48 output.
|
|
* [31] 0 = SRCMulti_SPDIF input from fxengine 0x60-0x67.
|
|
* 1 = SRCMulti_SPDIF input from SRC48 output.
|
|
*/
|
|
/* 0xffffffff -> C00100FF */
|
|
/* 0 -> Not playback sound, irq still running */
|
|
/* 0xXXXXXX10 = PCM0 Left/Right On. (Front)
|
|
* 0xXXXXXX20 = PCM1 Left/Right On. (Center/LFE)
|
|
* 0xXXXXXX40 = PCM2 Left/Right On. (Unknown)
|
|
* 0xXXXXXX80 = PCM3 Left/Right On. (Rear)
|
|
*/
|
|
#define PLAYBACK_SPDIF_SELECT 0x70 /* Default: 12030F00 */
|
|
/* 0xffffffff -> 3FF30FFF */
|
|
/* 0x00000001 pauses stream/irq fail. */
|
|
/* All other bits do not effect playback */
|
|
#define PLAYBACK_SPDIF_SRC_SELECT 0x71 /* Default: 0000E4E4 */
|
|
/* 0xffffffff -> F33FFFFF */
|
|
/* All bits do not effect playback */
|
|
#define PLAYBACK_SPDIF_USER_DATA0 0x72 /* SPDIF out user data 0 */
|
|
#define PLAYBACK_SPDIF_USER_DATA1 0x73 /* SPDIF out user data 1 */
|
|
/* 0x74-0x75 unknown */
|
|
#define CAPTURE_SPDIF_CONTROL 0x76 /* SPDIF in control setting */
|
|
#define CAPTURE_SPDIF_STATUS 0x77 /* SPDIF in status */
|
|
#define CAPURE_SPDIF_USER_DATA0 0x78 /* SPDIF in user data 0 */
|
|
#define CAPURE_SPDIF_USER_DATA1 0x79 /* SPDIF in user data 1 */
|
|
#define CAPURE_SPDIF_USER_DATA2 0x7a /* SPDIF in user data 2 */
|
|
|